1. Field of Invention
The present invention generally relates to data communication protocols, and more particularly, to systems and methods for quality of service management for multiple connections within a network communication system.
2. Description of Related Art
Transport Control Protocol (TCP) has become a common end-to-end data transport protocol used in modern data communication networks. Communication networks employing a TCP architecture offer significant advantages in terms of connectivity by enabling applications and users deployed on different physical networks to communicate with one another using a common communications protocol. The recent increase in the number and diversity of applications, users and networking environments utilizing TCP architectures, however, has exposed many of the limitations associated with a single, ubiquitous design. Because these architectures were primarily intended to provide reliable, sequenced transmission of non-real-time data streams over relatively high bandwidth wireline channels, these TCP architectures tend to exhibit sub-optimal performance when employed in applications or networking environments having different or incompatible characteristics.
Many of the problems associated with conventional TCP architectures stem from the flow control, congestion control and error recovery mechanisms used to control transmission of data over a communication network. Typical TCP flow control mechanisms, for example, utilize an acknowledgement-based approach to control the number and timing of new packets transmitted over the communication network. In these implementations, a sender maintains a congestion window parameter that specifies the maximum number of unacknowledged packets that may be transmitted to the receiver. As the sender receives acknowledgement signals from the receiver, the congestion control mechanism increases the size of the congestion window (and decreases the number of unacknowledged packets), thereby enabling the flow control mechanism to immediately transmit additional packets to the receiver. A problem with this approach is that it assumes that the network employs symmetric communication channels that enable data packets and acknowledgements to be equally spaced in time. In communication networks, such as wireless communication networks, that employ asymmetric uplink and downlink channels, where the available bandwidth towards the receiver is significantly higher than the available bandwidth towards the sender, the receiver may be unable to access the uplink channel in order to transmit acknowledgement signals to the sender in a timely manner. This initial delay in the transmission of acknowledgement signals may cause the sender to suspend transmission of additional data packets until additional acknowledgement signals are received, and then transmit a large burst of packets in response to the sender receiving a large group of acknowledgement signals. This bursty nature of data transmission may under-utilize the available bandwidth on the downlink channel, and may cause some applications requiring a steady flow of data, such as audio or video, to experience unusually poor performance.
The congestion control and error recovery mechanisms typically employed in TCP architectures may also cause the communication network to exhibit sub-optimal performance. In conventional TCP implementations, the congestion control and error recovery mechanisms are used to adjust the size of the congestion window (and therefore the number of new packets that may be transmitted to the receiver) based on the current state of the congestion control and error recovery algorithm. In the initial “slow start” state, for example, the sender rapidly probes for bandwidth by increasing the size of the congestion window by one for each new acknowledgement received from the receiver until the congestion window exceeds a certain congestion window threshold. Once the congestion window exceeds the congestion window threshold, the algorithm enters a “congestion avoidance” state, where the congestion window is increased by one whenever a number of acknowledgment signals equal to the size of the current congestion window is received. If the sender receives a predetermined number of duplicate acknowledgements or a selective acknowledgment (“SACK”) that indicate that a packet in the sequence has not been received, the algorithm enters a “fast retransmit” state in which the sender decreases the congestion window to a size equal to one half of the current congestion window plus three, and retransmits the lost packet. After the “fast retransmit” state, the algorithm enters a temporary “fast recovery” state that increments the congestion window by one for each duplicate acknowledgement received from the receiver. If an acknowledgement for the lost packet is received before a retransmit timeout occurs (which is typically based on the average and mean deviation of round-trip time samples), the algorithm transitions to the “congestion avoidance” state. On the other hand, if an acknowledgement for the lost packet is not received before a retransmit timeout occurs, the sender resets the congestion window to one, retransmits the lost packet and transitions to the “slow start” state.
The problem with the foregoing approach is that the congestion avoidance and error recovery mechanisms assume that packet loss within the communication network was caused by congestion, rather than a temporary degradation in the signal quality of the communication channel. Although this assumption may work adequately for many wireline communication networks that have a relatively low occurrence of random packet loss, random packet loss due to fading, temporary degradation in signal quality, signal handoffs or large propagation delays occur with relatively high frequency in most wireless and other bandwidth constrained networks. Because conventional TCP architectures react to both random loss and network congestion by significantly and repeatedly reducing the congestion window, high levels of random packet loss may lead to significant and potentially unjustified deterioration in data throughput. TCP performance, particularly in the fast recovery state, may also be adversely impacted by signal handoffs and fades that typically occur in wireless networks. Handoffs and fades can cause multiple data packet losses, which can lead to failure of TCP's fast recovery mechanism and result in prolonged timeouts. If the handoff or fade lasts for several round trip times, failure of multiple retransmission attempts may cause exponential backoff of data throughput. This may result in long recovery times that last significantly longer than the originating fades or handoffs, and may cause TCP connections to stall for extended periods of time.
The problems associated with conventional TCP architectures become especially apparent in situations involving multiple connections between a given sender and a given receiver. Many applications, such as web browsers, often open multiple TCP connections between a sender and a receiver so that data may be communicated in parallel. Under conventional TCP architectures, these connections operate independently and may compete with one another for the same bandwidth, even though these connections serve the same host or the same application. This may lead to inefficient use of resources with decreased overall throughput as each connection attempts to maximize its bandwidth without regard to other connections. For example, when a new connection is initiated between a sender and receiver, the TCP congestion control mechanism aggressively increases the size of the congestion window until it senses a data packet loss. This process may adversely impact other connections that share the same reduced-bandwidth channel as the connection being initialized attempts to maximize its data throughput without regard of the other pre-existing connections. Furthermore, because conventional TCP architectures do not distinguish between data packets communicated over each connection, the competition among connections may cause lower priority data, such as email data, to obtain a greater portion of the available bandwidth than higher priority data, such as real-time voice or video. This lack of coordination between multiple connections to the same host may produce a sub-optimal allocation of the available bandwidth as connections carrying low priority data consume the available bandwidth at the expense of connections carrying higher priority data.
Therefore, in light of the deficiencies of existing approaches, there is a need for improved systems and methods for quality of service management for multiple connections within a network communication system, particularly network communication systems having wireless and other bandwidth constrained channels.